Docstoc

Apparatus And Method Of Processing An Audio Signal - Patent 7987008

Document Sample
Apparatus And Method Of Processing An Audio Signal - Patent 7987008 Powered By Docstoc
					


United States Patent: 7987008


































 
( 1 of 1 )



	United States Patent 
	7,987,008



 Liebchen
 

 
July 26, 2011




Apparatus and method of processing an audio signal



Abstract

 In one embodiment, the method includes receiving the audio signal
     including at least one block of audio data and configuration information,
     and reading coding type information from the configuration information.
     The coding type information indicates an entropy coding scheme used in
     encoding the audio signal. Partitioning information is read from the
     configuration information, and the partitioning information indicates
     whether the block is divided into sub-blocks. The partitioned sub-blocks
     are decoded based on the entropy coding scheme if the block is divided
     into sub-blocks.


 
Inventors: 
 Liebchen; Tilman (Berlin, DE) 
 Assignee:


LG Electronics Inc.
 (Seoul, 
KR)





Appl. No.:
                    
12/232,743
  
Filed:
                      
  September 23, 2008

 Related U.S. Patent Documents   
 

Application NumberFiling DatePatent NumberIssue Date
 11481927Jul., 2006
 60697551Jul., 2005
 60700570Jul., 2005
 

 
Foreign Application Priority Data   
 

Jul 16, 2005
[WO]
PCT/KR2005/002290

Jul 16, 2005
[WO]
PCT/KR2005/002291

Jul 16, 2005
[WO]
PCT/KR2005/002292

Jul 18, 2005
[WO]
PCT/KR2005/002306

Jul 18, 2005
[WO]
PCT/KR2005/002307

Jul 18, 2005
[WO]
PCT/KR2005/002308



 



  
Current U.S. Class:
  700/94
  
Current International Class: 
  G06F 17/00&nbsp(20060101)
  
Field of Search: 
  
  
 700/94
  

References Cited  [Referenced By]
U.S. Patent Documents
 
 
 
4110571
August 1978
Hills

4922537
May 1990
Frederiksen

5089818
February 1992
Mahieux et al.

5166686
November 1992
Sugiyama

5243686
September 1993
Tokuda et al.

5283780
February 1994
Schuchmann et al.

5394473
February 1995
Davidson

5751773
May 1998
Campana, Jr.

5828784
October 1998
Miyashita et al.

5864801
January 1999
Sugiyama et al.

5915066
June 1999
Katayama

6041302
March 2000
Bruekers

6052661
April 2000
Yamaura et al.

6069947
May 2000
Evans et al.

6098039
August 2000
Nishida

6104996
August 2000
Yin

6124895
September 2000
Fielder

6154549
November 2000
Arnold et al.

6226608
May 2001
Fielder et al.

6278900
August 2001
Aihara

6300888
October 2001
Chen et al.

6366960
April 2002
Hawkes

6420980
July 2002
Ejima

6628714
September 2003
Fimoff et al.

6675148
January 2004
Hardwick

6678332
January 2004
Gardere et al.

6690307
February 2004
Karczewicz

6691082
February 2004
Aguilar et al.

6696993
February 2004
Karczewicz

6775254
August 2004
Willenegger et al.

6813602
November 2004
Thyssen

6816491
November 2004
Fujii et al.

6950603
September 2005
Isozaki et al.

6970479
November 2005
Abrahamsson et al.

7003451
February 2006
Kjorling et al.

7069223
June 2006
Matsumoto et al.

7085401
August 2006
Averbuch et al.

7096481
August 2006
Forecast et al.

7145484
December 2006
Moriya et al.

7203638
April 2007
Jelinek et al.

7266501
September 2007
Saunders et al.

7292902
November 2007
Smithers et al.

7392195
June 2008
Fejzo

7542905
June 2009
Kondo

2001/0025358
September 2001
Eidson et al.

2002/0049586
April 2002
Nishio et al.

2002/0165710
November 2002
Ojanpera

2003/0033569
February 2003
Middelink et al.

2003/0078687
April 2003
Du Breuil

2003/0115051
June 2003
Chen et al.

2003/0125933
July 2003
Saunders et al.

2003/0138157
July 2003
Schwartz

2004/0037461
February 2004
Lainema

2004/0049379
March 2004
Thumpudi et al.

2004/0076333
April 2004
Zhang et al.

2004/0117044
June 2004
Konetski

2004/0161116
August 2004
Tsuji et al.

2004/0196913
October 2004
Chakravarthy et al.

2004/0247035
December 2004
Schroder et al.

2005/0015259
January 2005
Thumpudi et al.

2005/0063368
March 2005
Reznik

2005/0071027
March 2005
Prakash et al.

2005/0080829
April 2005
Reznik

2005/0149322
July 2005
Bruhn et al.

2005/0152557
July 2005
Sasaki et al.

2005/0216262
September 2005
Fejzo

2005/0254281
November 2005
Sawabe et al.

2006/0013405
January 2006
Oh et al.

2006/0174267
August 2006
Schmidt

2006/0251330
November 2006
Toth et al.

2008/0075153
March 2008
Roberts et al.

2009/0030700
January 2009
Liebchen



 Foreign Patent Documents
 
 
 
1195940
Oct., 1998
CN

0 599 825
Jun., 1994
EP

691751
Jan., 1996
EP

1 047 047
Oct., 2000
EP

1054575
Nov., 2000
EP

1 074 976
Feb., 2001
EP

1359755
Nov., 2003
EP

1391880
Feb., 2004
EP

1 427 252
Jun., 2004
EP

2 318 029
Apr., 1998
GB

08-031096
Feb., 1996
JP

2004-054156
Feb., 2004
JP

2004-072345
Mar., 2004
JP

WO 92/12607
Jul., 1992
WO

WO 99/53479
Oct., 1999
WO

WO 00/41313
Jul., 2000
WO

WO 00/45389
Aug., 2000
WO

WO 03/049081
Jun., 2003
WO

WO 03/085645
Oct., 2003
WO

WO 2004-047305
Jun., 2004
WO

WO 2005/036529
Apr., 2005
WO



   
 Other References 

Office Action dated Dec. 18, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,933. cited by other
.
Office Action dated Dec. 24, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,941. cited by other
.
US Office Action dated Feb. 23, 2010 in corresponding U.S. Appl. No. 12/232,527. cited by other
.
ISO/IEC JTC1/SC29/WG11/N7016, "Text of ISO/IEC 14496-3:2001/FPDAM 4, Audio Lossless Coding (ALS), new audio profiles and BSAC extensions," Jan. 2005, Hong Kong, China. cited by other
.
Search Report dated Jun. 2, 2010 by the European Patent Office for Application No. 06769225.1. cited by other
.
Liebchen, T, "Proposed Text of ISO/IEC 14496-3:2001/FPDAM 4, Audio Lossless Coding (ALS), new audio profiles and BSAC"--Hong Kong, Jan. 2005. cited by other
.
Liebchen T. "Proposed Study on ISO/IEC 14496-3:2001/FPDAM 4, Audio Lossless Coding (ALS), new audio profiles and BSAC"--Busan, Korea, Apr. 2005. cited by other
.
Office Action dated Jul. 30, 2010 from the Chinese Patent Office for Application No. 2006-80029407. cited by other
.
Liebchen, T. "MPEG-4 Lossless Coding for High-Definition Audio", Audio Engineering Society Convention Paper, Oct. 2003, New York. cited by other
.
Liebchen, T. "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", Audio Engineering Society Convention Paper, May 2005, Barcelona. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002683 dated Oct. 16, 2006. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002687 dated Oct. 16, 2006. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002688 dated Oct. 16, 2006. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002685 dated Nov. 24, 2006. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002679 dated Jan. 4, 2007. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002690 dated Jan. 12, 2007. cited by other
.
International Search Report corresponding to Korean Application No. PCT/KR2006/002686 dated Jan. 18, 2007. cited by other
.
"An Introduction to MPEG-4 Audio Lossless Coding", Liebchen, T.; Acoustics, Speech, and Signal Processing, 2004. Proceedings. IEEE International Conference on vol. 3, May 17-21, 2004. pp. 1012-1015. cited by other
.
"MPEG-4 Lossless Coding for High-Definition Audio", Liebchen, T.; Audio Engineering Society Convention Paper, New York, NY, US. Oct. 10, 2003. Figure 3 and Chapter 3.4. cited by other
.
"MPEG Audio Layer II: A Generic Coding Standard for Two and Multichannel Sound for DVB, DAB and Computer Multimedia", Stoll, G.; International Broadcasting Convention, 1995 Amsterdam. pp. 136-144. cited by other
.
"Linear Prediction From Subbands for Lossless Audio Compression", Giurcaneanu C.D., Tabus I., Astola J.; NORSIG '98. 3.sup.rd IEEE NORDIC Signal Processing Symposium. Vigso, Denmark Jun. 8-11, 1998. pp. 225-228. cited by other
.
Office Action dated Oct. 20, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,932. cited by other
.
Office Action dated Oct. 27, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,926. cited by other
.
Office Action dated Sep. 14, 2009 by Japanese Patent Office for counterpart Japanese Application No. 2008-521316 (without translation). cited by other
.
ID3v2 Tag Specification, Mar. 1998. cited by other
.
ISO/IEC JTC 1/SC 29/WG11N6435 "WD3 of ISO/IEC 14496-3:2001/AMD 4, Audio Lossless Coding (ALS)" Mar. 2004, Munich, Germany. cited by other
.
Yamaha DME Designer Manual, Version 2.0, Copyright 2004. cited by other
.
T.H. Tsai, et al., "A Novel MPEG-2 Audio Decoder with Efficient Data Arrangement and Memory Configuration," IEEE Transactions on Consumer Electronics, 43(3): 598-604, (1997). cited by other
.
Kamamoto, Yutaka, et al., "Lossless Compression of Multi-channel Signals Using Inter-channel Correlation," Transactions of Information Processing Society of Japan, 46(5): 1118-1128, (2005). (English Abstract). cited by other
.
International Search Report dated Jan. 4, 2007. cited by other
.
International Search Report dated Jan. 18, 2007. cited by other
.
International Search Report dated Feb. 2, 2007. cited by other
.
International Search Report dated Jan. 10, 2007. cited by other
.
Office Action dated Aug. 3, 2010 from the USPTO for U.S. Appl. No. 12/232,526. cited by other
.
US Office Action dated Mar. 29, 2010 in corresponding U.S. Appl. No. 11/481,939. cited by other
.
US Office Action dated Apr. 1, 2010 for corresponding U.S. Appl. No. 12/232,662. cited by other
.
US Office Action dated Apr. 8, 2010 for corresponding U.S. Appl. No. 11/481,932. cited by other
.
US Office Action dated Mar. 5, 2010 in corresponding U.S. Appl. No. 11/481,915. cited by other
.
US Office Action dated Jan. 15, 2010 in corresponding U.S. Appl. No. 12/232,783. cited by other
.
US Office Action dated Jan. 15, 2010 in corresponding U.S. Appl. No. 11/481,932. cited by other
.
Office Action dated Nov. 20, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,942. cited by other
.
Office Action dated Dec. 1, 2009 by USPTO for counterpart U.S. Appl. No. 12/232,527. cited by other
.
Office Action dated Dec. 9, 2009 by USPTO for counterpart U.S. Appl. No. 11/481,930. cited by other
.
U.S. Office Action dated Oct. 7, 2010. cited by other
.
Office Action dated Jul. 30, 2010 from the Chinese Patent Office for Application No. 2006-80029407. cited by other
.
Liebchen, T. "MPEG-4 Lossless Coding for High-Definition Audio", Audio Engineering Society Convention Paper, Oct. 2003, New York. cited by other
.
Liebchen, T. "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", Audio Engineering Society Convention Paper, May 2005, Barcelona. cited by other
.
Office Action dated Aug. 6, 2010 from the Chinese Patent Office for Application No. 2006-80025139.5. cited by other
.
US Office Action dated Nov. 17, 2010 for co-pending U.S. Appl. No. 11/481,915. cited by other
.
US Notice of Allowance dated Nov. 22, 2010 for co-pending U.S. Appl. No. 11/481,939. cited by other
.
Liebchen et al., "MPEG-4 ALS: an Emerging Standard for Lossless Audio Coding," ISO/IEC, DCC, 2004 (10 pages). cited by other
.
Liebchen et al., "Audio Engineering Society Convention Paper, MPEG-4 Audio Lossless Coding," Presented at the 116.sup.th Convention, May 8-11, 2004, Berlin, Germany, pp. 1-9. cited by other
.
West et al., "Finding an Optimal Segmentation for Audio Genre Classification," ISMIR, 2005 (8 pages). cited by other
.
European Search Report dated Sep. 29, 2010 for U.S. Appl. No. 06769223.6. cited by other
.
D Yang, et al "A Lossless Audio Compression Scheme with Random Access Property"--IEEE International Conference--Montreal, Canada May 2004. cited by other
.
US Office Action dated Dec. 7, 2010, issued in co-pending U.S. Appl. No. 12/232,734. cited by other
.
Office Action dated Aug. 12, 2010 from the Chinese Patent Office for Application No. 2008-80030511.1 with English Translation. cited by other
.
Johnson, J.D. et al Article "Sum-Difference Stereo Transform Coding"--IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 2, pp. 569-572, Copyright Sep. 1992. cited by other
.
US Notice of Allowance dated Mar. 2, 2011, issued in co-pending U.S. Appl. No. 12/232,527. cited by other
.
European Office Action dated Apr. 20, 2011 for EP application No. 06 757 768.4. cited by other
.
"Study on 14496-3:2001/FPDAM 4, ALS", ITU Study Group 16--Video Coding Experts Group--ISO/IEC MPEG & ITU-T VCEG(ISO/IEC JTC1/SC29/WG 1 1 and ITU-T SG16 16).89 pages, No. N7134 (May 6, 2005). cited by other.  
  Primary Examiner: Kuntz; Curtis


  Assistant Examiner: McCord; Paul


  Attorney, Agent or Firm: Harness, Dickey & Pierce, P.L.C.



Parent Case Text



DOMESTIC PRIORITY INFORMATION


 This application is a continuation of co-pending application Ser. No.
     11/481,927 filed on Jul. 7, 2006, which claims the benefit of priority on
     U.S. Provisional Application Nos. 60/697,551 and 60/700,570 filed Jul.
     11, 2005 and Jul. 19, 2005, respectively; the entire contents of all of
     which are hereby incorporated by reference.


FOREIGN PRIORITY INFORMATION


 This application claims the benefit of priority on International PCT
     Application Nos. PCT/KR2005/002290, PCT/KR2005/002291, PCT/KR2005/002292,
     PCT/KR2005/002306, PCT/KR2005/002307 and PCT/KR2005/002308 filed Jul. 16,
     2005, Jul. 16, 2005, Jul. 16, 2005, Jul. 18, 2005, Jul. 18, 2005 and Jul.
     18, 2005, respectively, via the claim for priority on co-pending
     application Ser. No. 11/481,927; the entire contents of which are hereby
     incorporated by reference.

Claims  

I claim:

 1.  A method of decoding an audio signal by a decoder operated by a central processing unit, the method comprising: receiving, by the decoder, the audio signal including at least one
block of audio data and configuration information;  reading, by the decoder, coding type information from the configuration information, the coding type information indicating a Block Gilbert Moore Code (BGMC) scheme used in encoding the audio signal; 
reading, by the decoder, partitioning information from the configuration information, the partitioning information indicating a sub-block partitioning scheme, partitioning, by the decoder, at least one block of the audio data into sub-blocks based on the
sub-block partitioning scheme;  and decoding, by the decoder, the sub-blocks based on the entropy coding scheme and a number of the sub-blocks if the block is divided into sub-blocks, wherein an entropy coding scheme is one of a Rice code scheme and the
BGMC scheme, and if the entropy coding scheme is the BGMC scheme, the sub-block partitioning scheme is one of (i) a first partition scheme providing for selecting between no partitioning and partitioning the block into four sub-blocks, and (ii) a second
partition scheme providing for selecting between no partitioning, partitioning the block into two sub-blocks, partitioning the block into four sub-blocks and partitioning the block into eight sub-blocks, wherein the number of sub-blocks is determined
based on the entropy coding scheme and the sub-block partitioning scheme.


 2.  The method of claim 1, wherein if the entropy coding scheme is the Rice code scheme, the sub-block partition scheme is one of (i) a no partition scheme, and (ii) a partition scheme providing for selecting between no partitioning and
partitioning the block into four sub-blocks.


 3.  A method of processing an audio signal by an encoder operated by a central processing unit, the audio signal having at least one frame, the frame having at least one channel divided into a number of blocks, the method comprising: adding, by
the encoder, coding type information to configuration information for the audio signal, the coding type information indicating a Block Gilbert Moore Code (BGMC) scheme used in encoding the audio signal;  and adding, by the encoder, partitioning
information to the configuration information, the partitioning information indicating whether the block is divided into sub-blocks and indicating a sub-block partitioning scheme used to partition the blocks if the block is divided into sub-blocks,
wherein a different entropy coding scheme is one of a Rice code scheme and the BGMC scheme, and if the entropy coding scheme is the BGMC scheme, the sub-block partitioning scheme is one of (i) a first partition scheme providing for selecting between no
partitioning and partitioning the block into four sub-blocks, and (ii) a second partition scheme providing for selecting between no partitioning, partitioning the block into two sub-blocks, partitioning the block into four sub-blocks and partitioning the
block into eight sub-blocks, wherein a number of the sub-blocks is selected based on the entropy coding scheme and the sub-block partitioning scheme.


 4.  The method of claim 3, wherein if the entropy coding scheme is the Rice code scheme, the sub-block partition scheme is one of (i) a no partition scheme, and (ii) a partition scheme providing for selecting between no partitioning and
partitioning the block into four sub-blocks.


 5.  An apparatus of decoding an audio signal, the apparatus comprising: a central processing unit configured to operate a decoder, the decoder including, a demultiplexing part configured to receive the audio signal including at least one block
of audio data and configuration information, the demultiplexing part configured to read coding type information from the configuration information, the coding type information indicating a Block Gilbert Moore Code (BGMC) scheme used in encoding the audio
signal, read partitioning information from the configuration information, the partitioning information indicating a sub-block partitioning scheme, and partition at least one block of the audio data into sub-blocks based on the sub-block partitioning
scheme, and an entropy decoding part configured to decode the partitioned sub-blocks based on the entropy coding scheme and a number of the sub-blocks if the block is divided into sub-blocks, wherein an entropy coding scheme is one of a Rice code scheme
and the BGMC scheme, and if the entropy coding scheme is the BGMC scheme, the sub-block partitioning scheme is one of (i) a first partition scheme providing for selecting between no partitioning and partitioning the block into four sub-blocks, and (ii) a
second partition scheme providing for selecting between no partitioning, partitioning the block into two sub-blocks, partitioning the block into four sub-blocks and partitioning the block into eight sub-blocks, wherein the number of sub-blocks is
determined based on the entropy coding scheme and the sub-block partitioning scheme.


 6.  The apparatus of claim 5, wherein if the entropy coding scheme is the Rice code scheme, the sub-block partition scheme is one of (i) a no partition scheme, and (ii) a partition scheme providing for selecting between no partitioning and
partitioning the block into four sub-blocks.


 7.  An apparatus for processing an audio signal having at least one frame, the frame having at least one channel divided into a number of blocks, the apparatus comprising: a central processing unit configured to operate an encoder, the encoder
including, a partitioning part configured to generate coding type information to configuration information for the audio signal, the coding type information indicating a Block Gilbert Moore Code (BGMC) scheme used in encoding the audio signal, and
generate partitioning information to the configuration information, wherein the partitioning information indicates whether the block is divided into sub-blocks and further indicates a sub-block partitioning scheme used to partition the blocks if the
block is divided into sub-blocks, wherein an entropy coding scheme is one of a Rice code scheme and the BGMC scheme, and if the entropy coding scheme is the BGMC scheme, the sub-block partitioning scheme is one of (i) a first partitioning scheme
providing for selecting between no partitioning and partitioning the block into four sub-blocks, and (ii) a second partition scheme providing for selecting between no partitioning, partitioning the block into two sub-blocks, partitioning the block into
four sub-blocks and partitioning the block into eight sub-blocks, wherein a number of the sub-blocks is selected based on the entropy coding scheme and the sub-block partitioning scheme.


 8.  The apparatus of claim 7, wherein if the entropy coding scheme is the Rice code scheme, the sub-block partition scheme is one of (i) a no partition scheme, and (ii) a partition scheme providing for selecting between no partitioning and
partitioning the block into four sub-blocks.  Description  

BACKGROUND OF THE INVENTION


 The present invention relates to a method for processing audio signal, and more particularly to a method and apparatus of encoding and decoding audio signal.


 The storage and replaying of audio signals has been accomplished in different ways in the past.  For example, music and speech have been recorded and preserved by phonographic technology (e.g., record players), magnetic technology (e.g.,
cassette tapes), and digital technology (e.g., compact discs).  As audio storage technology progresses, many challenges need to be overcome to optimize the quality and storability of audio signals.


 For the archiving and broadband transmission of music signals, lossless reconstruction is becoming a more important feature than high efficiency in compression by means of perceptual coding as defined in MPEG standards such as MP3 or AAC. 
Although DVD audio and Super CD Audio include proprietary lossless compression schemes, there is a demand for an open and general compression scheme among content-holders and broadcasters.  In response to this demand, a new lossless coding scheme has
been considered as an extension to the MPEG-4 Audio standard.  Lossless audio coding permits the compression of digital audio data without any loss in quality due to a perfect reconstruction of the original signal.


SUMMARY OF THE INVENTION


 The present invention relates to method of processing an audio signal.


 In one embodiment, at least one audio data frame having at least one channel is generated.  Each channel is divided into a plurality of blocks.  A sub-block partitioning scheme is selected, and a number of sub-blocks into which the block is to
be partitioned is selected.  The selected number of sub-blocks is chosen from numbers of sub-blocks available for the selected sub-block partitioning scheme.  The block of audio data is partitioned into sub-blocks according to the selected number of
sub-blocks, and the partitioned sub-blocks are coded according to a selected entropy coding scheme.


 In one embodiment, the selected entropy coding scheme is one of a Rice code scheme and a Block Gilbert Moore Code (BGMC) scheme.


 In one embodiment, the method further includes determining a plurality of code parameters s(0), s(1), .  . . s(N-1), entropy coding the N sub-blocks using a plurality of entropy codes defined by the plurality of code parameters, and transmitting
the code parameters.  The transmitting may include directly transmitting s(0) representing the code parameter of the first sub-block, and transmitting a difference s(i)-s(i-1) for i=1, .  . . N-1, s(i) representing the code parameter of i.sub.th
sub-block following the first sub-block.


 Another embodiment is directed to a method of processing an audio signal having at least one frame.  The frame has at least one channel divided into a number of blocks.  The method may include adding coding information to configuration
information for the audio signal.  The coding information indicates an entropy coding scheme used in encoding the audio signal.  The method may include adding partition information to the configuration information.  The partition information indicates a
sub-block partitioning scheme used to partition the blocks.  The method may further include_adding sub-block information to at least one of the blocks.  The sub-block information indicates a number of sub-blocks into which the block is partitioned.


 In one embodiment, a number of bits in the sub-block information varies.  For example, the number of bits in the sub-block information varies depending on the sub-block partitioning scheme and the entropy coding scheme.


 In one embodiment, the entropy coding scheme is one of a Rice code scheme and a Block Gilbert Moore Code (BGMC) scheme.


 In another embodiment, if the entropy coding scheme is the Rice code scheme, the sub-block partition scheme is one of (i) a no partition scheme, and (ii) a partitioning scheme providing for selecting between no partitioning and partitioning the
block into four sub-blocks.


 If the entropy coding scheme is the BGMC scheme, in one embodiment, the sub-block partition scheme is one of (i) a first partitioning scheme providing for selecting between no partitioning and partitioning the block into four sub-blocks, and
(ii) a second partitioning scheme providing for selecting between no partitioning, partitioning the block into two sub-blocks, partitioning the block into four sub-blocks and partitioning the block into eight sub-blocks.


 In yet another embodiment, an audio signal including at least one block of audio data and configuration information is received.  Partitioning information is read from the configuration information, and the partitioning information indicates a
sub-block partitioning scheme by which the block is divided into sub-blocks.  Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the
sub-block partitioning scheme.  The partitioned sub-blocks are decoded based on the number of the sub-blocks.


 In one embodiment, the method further includes reading coding information from the configuration information for the audio signal.  Here, the coding information indicates an entropy coding scheme used in encoding the audio signal.  The
partitioned sub-blocks are decoded based on the number of the sub-blocks and the entropy coding scheme.


 In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information from the configuration information.  The coding type information
indicates an entropy coding scheme used in encoding the audio signal.  Partitioning information is read from the configuration information, and the partitioning information indicates whether the block is divided into sub-blocks.  The partitioned
sub-blocks are decoded based on the entropy coding scheme if the block is divided into sub-blocks.


 The present invention further relates to methods and apparatuses for encoding an audio signal, and to methods and apparatuses for decoding an audio signal. 

BRIEF DESCRIPTION OF THE DRAWINGS


 The accompanying drawings, which are included to provide a further understanding of the invention and are incorporated in and constitute a part of this application, illustrate embodiment(s) of the invention and together with the description
serve to explain the principle of the invention.  In the drawings:


 FIG. 1 is an example illustration of an encoder according to an embodiment of the present invention.


 FIG. 2 is an example illustration of a decoder according to an embodiment of the present invention.


 FIG. 3 is an example illustration of a bitstream structure of a compressed M-channel file according to an embodiment of the present invention.


 FIG. 4 is an example illustration of a conceptual view of a hierarchical block switching method according to an embodiment of the present invention.


 FIG. 5 is an example illustration of a block switching examples and corresponding block switching information codes.


 FIG. 6 is an example illustration of block switching methods for a plurality of channel according to embodiments of the present invention.


DETAILED DESCRIPTION OF EXAMPLE EMBODIMENTS


 Reference will now be made in detail to the preferred embodiments of the present invention, examples of which are illustrated in the accompanying drawings.  Wherever possible, the same reference numbers will be used throughout the drawings to
refer to the same or like parts.


 Prior to describing the present invention, it should be noted that most terms disclosed in the present invention correspond to general terms well known in the art, but some terms have been selected by the applicant as necessary and will
hereinafter be disclosed in the following description of the present invention.  Therefore, it is preferable that the terms defined by the applicant be understood on the basis of their meanings in the present invention.


 In a lossless audio coding method, since the encoding process has to be perfectly reversible without loss of information, several parts of both encoder and decoder have to be implemented in a deterministic way.


 Codec Structure


 FIG. 1 is an example illustration of an encoder 1 according to the present invention.


 A partitioning part 100 partitions the input audio data into frames.  Within one frame, each channel may be further subdivided into blocks of audio samples for further processing.  A buffer 110 stores block and/or frame samples partitioned by
the partitioning part 100.


 A coefficient estimating part 120 estimates an optimum set of coefficient values for each block.  The number of coefficients, i.e., the order of the predictor, can be adaptively chosen as well.  The coefficient estimating part 120 calculates a
set of parcor values for the block of digital audio data.  The parcor value indicates parcor representation of the predictor coefficient.  A quantizing part 130 quantizes the set of parcor values.


 A first entropy coding part 140 calculates parcor residual values by subtracting an offset value from the parcor value, and encodes the parcor residual values using entropy codes defined by entropy parameters, wherein the offset value and the
entropy parameters are chosen from an optimal table.  The optimal table is selected from a plurality of tables based on a sampling rate of the block of digital audio data.  The plurality of tables are predefined for a plurality of sampling rate ranges,
respectively, for optimal compression of the digital audio data for transmission.


 A coefficient converting part 150 converts the quantized parcor values into linear predictive coding (LPC) coefficients.  A predictor 160 estimates current prediction values from the previous original samples stored in the buffer 110 using the
linear predictive coding coefficients.  A subtractor 170 calculates a prediction residual of the block of digital audio data using an original value of digital audio data stored in the buffer 110 and a prediction value estimated in the predictor 160.


 A second entropy coding part 180 codes the prediction residual using different entropy codes and generates code indices.  The indices of the chosen codes will be transmitted as auxiliary information.  The second entropy coding part 180 may code
the prediction residual using one of two alternative coding techniques having different complexities.  One coding technique is the well-known Golomb-Rice coding (herein after simply "Rice code") method and the other is the well-known Block Gilbert-Moore
Codes (herein after simply "BGMC") method.  Rice codes have low complexity yet are efficient.  The BGMC arithmetic coding scheme offers even better compression at the expense of a slightly increased complexity compared to Rice codes.


 Finally, a multiplexing part 190 multiplexes coded prediction residual, code indices, coded parcor residual values, and other additional information to form a compressed bitstream.  The encoder 1 also provides a cyclic redundancy check (CRC)
checksum, which is supplied mainly for the decoder to verify the decoded data.  On the encoder side, the CRC can be used to ensure that the compressed data are losslessly decodable.


 Additional encoding options include flexible block switching scheme, random access and joint channel coding.  The encoder 1 may use these options to offer several compression levels with different complexities.  The joint channel coding is used
to exploit dependencies between channels of stereo or multi-channel signals.  This can be achieved by coding the difference between two channels in the segments where this difference can be coded more efficiently than one of the original channels.  These
encoding options will be described in more detail below after a description of an example decoder according to the present invention.


 FIG. 2 is an example illustration of a decoder 2 according to the present invention.  More specially, FIG. 2 shows the lossless audio signal decoder which is significantly less complex than the encoder, since no adaptation has to be carried out.


 A demultiplexing part 200 receives an audio signal and demultiplexes a coded prediction residual of a block of digital audio data, code indices, coded parcor residual values and other additional information.  A first entropy decoding part 210
decodes the parcor residual values using entropy codes defined by entropy parameters and calculates a set of parcor values by adding offset values to the decoded parcor residual values; wherein the offset value and the entropy parameters are chosen from
a table selected by the decoder from a plurality of tables based on a sampling rate of the block of digital audio data.  A second entropy decoding part 220 decodes the demultiplexed coded prediction residual using the code indices.  A coefficient
converting part 230 converts the entropy decoded parcor value into LPC coefficients.  A predictor 240 estimates a prediction residual of the block of digital audio data using the LPC coefficients.  An adder 250 adds the decoded prediction residual to the
estimated prediction residual to obtain the original block of digital audio data.  An assembling part 260 assembles the decoded block data into frame data.


 Therefore, the decoder 2 decodes the coded prediction residual and the parcor residual values, converts the parcor residual values into LPC coefficients, and applies the inverse prediction filter to calculate the lossless reconstruction signal. 
The computational effort of the decoder 2 depends on the prediction orders chosen by the encoder 1.  In most cases, real-time decoding is possible even on low-end systems.


 FIG. 3 is an example illustration of a bitstream structure of a compressed audio signal including a plurality of channels (e.g., M channels) according to the present invention.


 The bitstream consists of at least one audio frame including a plurality of channels (e.g., M channels).  The "channels" field in the bitstream configuration syntax (see Table 6 below) indicates the number of channels.  Each channel is
sub-divided into a plurality of blocks using the block switching scheme according to present invention, which will be described in detail later.  Each sub-divided block has a different size and includes coding data according to the encoding of FIG. 1. 
For example, the coding data within a subdivided block contains the code indices, the prediction order K, the predictor coefficients, and the coded residual values.  If joint coding between channel pairs is used, the block partition is identical for both
channels, and blocks are stored in an interleaved fashion.  A "js_stereo" field in the bitstream configuration syntax (Table 6) indicates whether joint stereo (channel difference) is on or off, and a "js_switch" field in the frame_data syntax (See Table
7 below) indicates whether joint stereo (channel difference) is selected.  Otherwise, the block partition for each channel is independent.


 Hereinafter, the block switching, random access, prediction, and entropy coding options previously mentioned will now be described in detail with reference to the accompanying drawings and syntaxes that follow.


 Block Switching


 An aspect of the present invention relates to subdividing each channel into a plurality of blocks prior to using the actual coding scheme.  Hereinafter, the block partitioning (or subdividing) method according to the present invention will be
referred to as a "block switching method".


 Hierarchical Block Switching


 FIG. 4 is an example illustration of a conceptual view of a hierarchical block switching method according to the present invention.  For example, FIG. 4 illustrates a method of hierarchically subdividing one channel into 32 blocks.  When a
plurality of channels is provided in a single frame, each channel may be subdivided (or partitioned) to up to 32 blocks, and the subdivided blocks for each channel configure a frame.


 Accordingly, the block switching method according to the present invention is performed by the partitioning part 100 shown in FIG. 1.  Furthermore, as described above, the prediction and entropy coding are performed on the subdivided block
units.


 In general, conventional Audio Lossless Coding (ALS) includes a relatively simple block switching mechanism.  Each channel of N samples is either encoded using one full length block (N.sub.B=N) or four blocks of length N.sub.B=N/4 (e.g., 1:4
switching), where the same block partition applies to all channels.  Under some circumstances, this scheme may have some limitations.  For example, while only 1:1 or 1:4 switching may be possible, different switching (e.g., 1:2, 1:8, and combinations
thereof) may be more efficient in some cases.  Also in conventional ALS, switching is performed identically for all channels, although different channels may benefit from different switching (which is especially true if the channels are not correlated).


 Therefore, the block switching method according to embodiments of the present invention provide relatively flexible block switching schemes, where each channel of a frame may be hierarchically subdivided into a plurality of blocks.  For example,
FIG. 4 illustrates a channel which can be hierarchically subdivided to up to 32 blocks.  Arbitrary combinations of blocks with N.sub.B=N, N/2, N/4, N/8, N/16, and N/32 may be possible within a channel according to the presented embodiments, as long as
each block results from a subdivision of a superordinate block of double length.  For example, as illustrated in the example shown in FIG. 4, a partition into N/4+N/4+N/2 may be possible, while a partition into N/4+N/2+N/4 may not be possible (e.g.,
block switching examples shown in FIGS. 5(e) and 5 described below).  Stated another way, the channel is divided into the plurality of blocks such that each block has a length equal to one of, N/(m.sup.i) for i=1, 2, .  . . p, where N is the length of
the channel, m is an integer greater than or equal to 2, and p represents a number of the levels in the subdivision hierarchy.


 Accordingly, in embodiments of the present invention, a bitstream includes information indicating block switching levels and information indicating block switching results.  Herein, the information related to block switching is included in the
syntax, which is used in the decoding process, described in detail below.


 For example, settings are made so that a minimum block size generated after the block switching process is N.sub.B=N/32.  However, this setting is only an example for simplifying the description of the present invention.  Therefore, settings
according to the present invention are not limited to this setting.


 More specifically, when the minimum block size is N.sub.B=N/32, this indicates that the block switching process has been hierarchically performed 5 times, which is referred to as a level 5 block switching.  Alternatively, when the minimum block
size is N.sub.B=N/16, this indicates that the block switching process has been hierarchically performed 4 times, which is referred to as a level 4 block switching.  Similarly, when the minimum block size is N.sub.B=N/8, the block switching process has
been hierarchically performed 3 times, which is referred to as a level 3 block switching.  And, when the minimum block size is N.sub.B=N/4, the block switching process has been hierarchically performed 2 times, which is referred to as a level 2 block
switching.  When the minimum block size is N.sub.B=N/2, the block switching process has been hierarchically performed 1 time, which is referred to as a level 1 block switching.  Finally, when the minimum block size is N.sub.B=N, the hierarchical block
switching process has not been performed, which is referred to as a level 0 block switching.


 In embodiments of the present invention, the information indicating the block switching level will be referred to as a first block switching information.  For example, the first block switching information may be represented by a 2-bit
"block_switching" field within the syntax shown in Table 6, which will be described in a later process.  More specifically, "block_switching=00" signifies level 0, "block_switching=01" signifies any one of level 1 to level 3, "block_switching=10"
signifies level 4, and "block_switching=11" signifies level 5.


 Additionally, information indicating the results of the block switching performed for each hierarchical level in accordance with the above-described block switching levels is referred to in the embodiments as second block switching information. 
Herein, the second block switching information may be represented by a "bs_info" field which is expressed by any one of 8 bits, 16 bits, and 32 bits within the syntax shown in Table 7.  More specifically, if "block_switching=01" (signifying any one of
level 1 to level 3), "bs_info" is expressed as 8 bits.  If "block_switching=10" (signifying level 4), "bs_info" is expressed as 16 bits.  In other words, up to 4 levels of block switching results may be indicated by using 16 bits.  Furthermore, if
"block_switching=11" (signifying level 5, "bs_info" is expressed as 32 bits.  In other words, up to 5 levels of block switching results may be indicated by using 32 bits.  Finally, if "block_switching=00" (signifying that the block switching has not been
performed), "bs_info" is not transmitted.  This signifies that one channel configures one block.


 The total number of bits being allocated for the second block switching information is decided based upon the level value of the first block switching information.  This may result in reducing the final bit rate.  The relation between the first
block switching information and the second block switching information is briefly described in Table 1 below.


 TABLE-US-00001 TABLE 1 Block switching levels.  Maximum #levels Minimum N.sub.B #Bytes for "bs_info" 0 N 0 ("block_switching = 00") 1 N/2 1 (=8bits) ("block_switching = 01") 2 N/4 1 (=8bits) ("block_switching = 01") 3 N/8 1 (=8bits)
("block_switching = 01") 4 N/16 2 (=16bits) ("block_switching = 10") 5 N/32 4 (=32bits) ("block_switching = 11")


 Hereinafter, an embodiment of a method of configuring (or mapping) each bit within the second block switching information (bs_info) will now be described in detail.


 The bs_info field may include up to 4 bytes in accordance with the above-described embodiments.  The mapping of bits with respect to levels 1 to 5 may be [(0) 1223333 44444444 55555555 55555555].  The first bit may be reserved for indicating
independent or synchronous block switching, which is described in more detail below in the Independent/Synchronous Block Switching section.  FIGS. 5(a)-5(f) illustrate different block switching examples for a channel where level 3 block switching may
take place.  Therefore, in these examples, the minimum block length is N.sub.B=N/8, and the bs_info consists of one byte.  Starting from the maximum block length N.sub.B=N, the bits of bs_info are set if a block is further subdivided.  For example, in
FIG. 5(a), there is no subdivision at all, thus "bs_info" is (0)000 0000.  In FIG. 5(b), the frame is subdivided ((0)1 .  . . ) and the second block of length N/2 is further split ((0)101 .  . . ) into two blocks of length N/4; thus "bs_info" is (0)1010
0000.  In FIG. 5(c), the frame is subdivided ((0)1 .  . . ), and only the first block of length N/2 is further split ((0)110 .  . . ) into two blocks of length N/4; thus "bs_info" is (0)1100 0000.  In FIG. 5(d), the frame is subdivided ((0)1 .  . . ),
the first and second blocks of length N/2 is further split ((0) 111 .  . . ) into two blocks of length N/4, and only the second block of length N/4 is further split ((0)11101 .  . . ) into two blocks of length N/8; thus "bs_info" is (0) 111 0100.


 As discussed above, the examples in FIGS. 5(e) and 5(f) represent cases of block switching that are not permitted because the N/2 block in FIG. 5(e) and the first N/4 block in FIG. 5(f) could not have been obtained by subdividing a block of the
previous level.


 Independent/Synchronous Block Switching


 FIGS. 6(a)-6(c) are example illustrations of block switching according to embodiments of the present invention.


 More specifically, FIG. 6(a) illustrates an example where block switching has not been performed for channels 1, 2, and 3.  FIG. 6(b) illustrates an example in which two channels (channels 1 and 2) configure one channel pair, and block switching
is performed synchronously in channels 1 and 2.  Interleaving is also applied in this example.  FIG. 6(c) illustrates an example in which two channels (channels 1 and 2) configure one channel pair, and the block switching of channels 1 and 2 is performed
independently.  Herein, the channel pair refers to two arbitrary audio channels.  The decision on which channels are grouped into channel pairs can be made automatically by the encoder or manually by the user.  (e.g., L and R channels, Ls and Rs
channels).


 In independent block switching, while the length of each channel may be identical for all channels, the block switching can be performed individually for each channel.  Namely, as shown in FIG. 6(c), the channels may be divided into blocks
differently.  If the two channels of a channel pair are correlated with each other and difference coding is used, both channels of a channel pair may be block switched synchronously.  In synchronous block switching, the channels are block switched (i.e.,
divided into blocks) in the same manner.  FIG. 6(b) illustrates an example of this, and further illustrates that the blocks may be interleaved.  If the two channels of a channel pair are not correlated with each other, difference coding may not provide a
benefit, and thus there will be no need to block switch the channels synchronously.  Instead, it may be more appropriate to switch the channels independently.


 Furthermore, according to another embodiment of the present invention, the described method of independent or synchronous block switching may be applied to a multi-channel group having a number of channels equal to or more than 3 channels.  For
example, if all channels of a multi-channel group are correlated with each other, all channels of a multi-channel group may be switched synchronously.  On the other hand, if all channels of a multi-channel group are not correlated with each other, each
channel of the multi-channel group may be switched independently.


 Moreover, the "bs_info" field is used as the information for indicating the block switching result.  Additionally, the "bs_info" field is also used as the information for indicating whether block switching has been performed independently or
performed synchronously for each channel configuring the channel pair.  In this case, as described above, a particular bit (e.g., first bit) within the "bs_info" field may be used.  If, for example, the two channels of the channel pair are independent
from one another, the first bit of the "bs_info" field is set to "1".  On the other hand, if the two channels of the channel pair are synchronous to one another, the first bit of the "bs_info" field is set as "0".


 Hereinafter, FIGS. 6(a), 6(b), and 6(c) will now be described in detail.


 Referring to FIG. 6(a), since none of the channels perform block switching, the related "bs_info" is not generated.


 Referring to FIG. 6(b), channels 1 and 2 configure a channel pair, wherein the two channels are synchronous to one another, and wherein block switching is performed synchronously.  For example, in FIG. 6(b), both channels 1 and 2 are split into
blocks of length N/4, both having the same bs_info "bs_info=(0)101 0000".  Therefore, one "bs_info" may be transmitted for each channel pair, which results in reducing the bit rate.  Furthermore, if the channel pair is synchronous, each block within the
channel pair may be required to be interleaved with one another.  The interleaving may be beneficial (or advantageous).  For example, a block of one channel (e.g., block 1.2 in FIG. 6(b)) within a channel pair may depend on previous blocks from both
channels (e.g., blocks 1.1 and 2.1 in FIG. 6(b)), and so these previous blocks should be available prior to the current one.


 Referring to FIG. 6(c), channels 1 and 2 configure a channel pair.  However, in this example, block switching is performed independently.  More specifically, channel 1 is split into blocks of a size (or length) of up to N/4 and has a bs_info of
"bs_info=(1)101 0000".  Channel 2 is split into blocks of a size of up to N/2 and has a bs_info of "bs_info=(1)100 0000".  In the example shown in FIG. 6(c), block switching is performed independently among each channel, and therefore, the interleaving
process between the blocks is not performed.  In other words, for the channel having the blocks switched independently, channel data may be arranged separately.


 Joint Channel Coding


 Joint channel coding, also called joint stereo, can be used to exploit dependencies between two channels of a stereo signal, or between any two channels of a multi-channel signal.  While it is straightforward to process two channels x.sub.1(n)
and x.sub.2 (n) independently, a simple method of exploiting dependencies between the channels is to encode the difference signal: d(n)=x.sub.2(n)-x.sub.1(n)


 instead of x1(n) or x2(n).  Switching between x.sub.1(n), x.sub.2(n) and d(n) in each block may be carried out by comparison of the individual signals, depending on which two signals can be coded most efficiently.  Such prediction with switched
difference coding is advantageous in cases where two channels are very similar to one another.  In case of multi-channel material, the channels can be rearranged by the encoder in order to assign suitable channel pairs.


 Besides simple difference coding, lossless audio codec also supports a more complex scheme for exploiting inter-channel redundancy between arbitrary channels of multi-channel signals.


 Random Access


 The present invention relates to audio lossless coding and is able to supports random access.  Random access stands for fast access to any part of the encoded audio signal without costly decoding of previous parts.  It is an important feature
for applications that employ seeking, editing, or streaming of the compressed data.  In order to enable random access, within a random access unit, the encoder needs to insert a frame that can be decoded without decoding previous frames.  The inserted
frame is referred to as a "random access frame".  In such a random access frame, no samples from previous frames may be used for prediction.


 Hereinafter, the information for random access according to the present invention will be described in detail.  Referring to the configuration syntax (shown in Table 6), information related with random access are transmitted as configuration
information.  For example, a "random_access" field is used as information for indicating whether random access is allowed, which may be represented by using 8 bits.  Furthermore, if random access is allowed, the 8-bit "random_access" field designates the
number of frames configuring a random access unit.  For example, when "random_access=0000 0000", the random access is not supported.  In other words, when "random_access>0", random access is supported.  More specifically, when "random_access=0000
0001,", this indicates that the number of frames configuring the random access unit is 1.  This signifies that random access is allowed in all frame units.  Furthermore, when "random_access=1111 1111", this indicates that the number of frames configuring
the random access unit is 255.  Accordingly, the "random_access" information corresponds to a distance between a random access frame within the current random access unit and a random access frame within the next random access unit.  Herein, the distance
is expressed by the number of frames.


 A 32-bit "ra_unit_size" field is included in the bitstream and transmitted.  Herein, the "ra_unit_size" field indicates the size of the random access unit in bytes; and therefore, indicates the distances from the current random access frame to
the next random access frame in bytes.  The "ra_unit_size" field is either included in the configuration syntax (Table 6) or included in the frame-data syntax (Table 7).  The configuration syntax (Table 6) may further include information indicating a
location where the "ra_unit_size" information is stored within the bitstream.  This information is represented as a 2-bit "ra_flag" field.  More specifically, for example, when "ra_flag=00", this indicates that the "ra_unit_size" information is not
stored in the bitstream.  When the "ra_flag=01", this indicates that the "ra_unit_size" information is stored in the frame-data syntax (Table 7) within the bitstream.  Furthermore, when the "ra_flag=10", the "ra_unit_size" information is stored in the
configuration syntax (Table 6) within the bitstream.  If the "ra_unit_size" information is included in the configuration syntax, this indicates that the "ra_unit_size" information is transmitted on the bitstream only one time and is applied equally to
all random access units.  Alternatively, if the "ra_unit_size" information is included in the frame-data syntax, this indicates the distance between the random access frame within the current random access unit and the random access frame within the next
random access unit.  Therefore, the "ra_unit_size" information is transmitted for each random access unit within the bitstream because the distance may change.


 Accordingly, the "random_access" field within the configuration syntax (Table 6) may also be referred to as first general information.  And, the "ra_flag" field may also be referred to as second general information.  In this aspect of the
present invention, an audio signal includes configuration information and a plurality of random access units, each random access unit containing one or more audio data frames, one of which is a random access frame, wherein the configuration information
includes first general information indicating a distance between two adjacent random access frames in frames, and second general information indicating where random access unit information for each random access unit is stored.  The random access unit
size information indicates a distance between two adjacent random access frames in bytes.


 Alternatively, in this aspect of the present invention, a method of decoding an audio signal includes receiving the audio signal having configuration information and a plurality of random access units, each random access unit containing one or
more audio data frames, one of which is a random access frame, reading first general information from the configuration information, the first general information indicating a distance between two adjacent random access frames in frames, and reading
second general information from the configuration information, the second general information indicating where random access size information for each random access unit is stored, and the random access unit size information indicating a distance between
two adjacent random access frames in bytes.  The decoder may then access the random access unit size information and use this and the first and second general information to perform random access of the audio data in the audio signal.


 Channel Configuration


 As shown in FIG. 3, an audio signal includes multi-channels information according to the present invention.  For example, each channel may be mapped at a one-to-one correspondence with a location of an audio speaker.  The configuration syntax
(Table 6 below) includes channel configuration information, which is indicated as a 16-bit "chan_config_info" field and a 16-bit "channels" field.  The "chan_config_info" field includes information for mapping the channels to the loudspeaker locations
and the 16-bit "channels" field includes information indicating the total number of channels.  For example, when the "channels" field is equal to "0", this indicates that the channel corresponds to a mono channel.  When the "channels" field is equal to
"1", this indicates that the channel corresponds to one of stereo channels.  And, when the "channels" field is equal to or more than "2", this indicates that the channel corresponds to one of multi-channels.


 Table 2 below shows examples of each bit configuring the "chan_config_info" field and each respective channel corresponding thereto.  More specifically, when a corresponding channel exists within the transmitted bitstream, the corresponding bit
within the "chan_config_info" field is set to "1".  Alternatively, when a corresponding channel does not exist within the transmitted bitstream, the corresponding bit within the "chan_config_info" field is set to "0".  The present invention also includes
information indicating whether the "chan_config_info" field exists within the configuration syntax (Table 6).  This information is represented as a 1-bit "chan_config" flag.  More specifically, "chan_config=0" indicates that the "chan_config_info" field
does not exist.  And, "chan_config=1" indicates that the "chan_config_info" field exists.  Therefore, when "chan_config=0", this indicates that the "chan_config_info" field is not newly defined within the configuration syntax (Table 6).


 TABLE-US-00002 TABLE 2 Channel configuration.  Bit position in Speaker location Abbreviation chan_config_info Left L 1 Right R 2 Left Rear Lr 3 Right Rear Rr 4 Left Side Ls 5 Right Side Rs 6 Center C 7 Center Rear/ S 8 Surround Low Frequency LFE
9 Effects Left Downmix L0 10 Right Downmix R0 11 Mono Downmix M 12 (reserved) 13-16


 Furthermore, channel configuration may be such that correlated channels are not adjoining in the structure of the audio data or audio signal.  Accordingly, it may be beneficial to rearrange the channel data to make correlated channels adjoining. The manner in which the channels are rearranged may be indicating in the chan_pos field of the configuration information (Table 6).  Further, the configuration information (Table 6) may include a chan_sort field indicating whether the channel
rearrangement information of the chan_pos field is present.


 Based on reading the various channel information discussed above, the decoder processes the audio signal so that the correct channel is sent to the correct speaker.


 Frame Length


 As shown in FIG. 3, an audio signal includes multiple or multi-channels according to the present invention.  Therefore, when performing encoding, information on the number of multi-channels configuring one frame and information on the number of
samples for each channel are inserted in the bitstream and transmitted.  Referring to the configuration syntax (Table 6), a 32-bit "samples" field is used as information indicating the total number of audio data samples configuring each channel. 
Further, a 16-bit "frame_length" field is used as information indicating the number of samples for each channel within the corresponding frame.


 Furthermore, a 16-bit value of the "frame_length" field is determined by a value used by the encoder, and is referred to as a user-defined value.  In other words, instead of being a fixed value, the user-defined value may be arbitrarily
determined upon the encoding process.  For example, the value may be set by a user of the encoding process.


 Therefore, during the decoding process, when the bitstream is received through the demultiplexing part 200 of shown in FIG. 2, the frame number of each channel should first be obtained.  This value is obtained according to the algorithm shown
below.


 TABLE-US-00003 frames = samples / frame_length; rest = samples % frame_length; if (rest) { frames++; frlen_last = rest; } else frlen_last = frame_length;


 More specifically, the total number of frames for each channel is calculated by dividing the total number of samples for each channel, which is decided by the "samples" field transmitted through the bitstream, by the number of samples within a
frame of each channel, which is decided by the "frame_length" field.  For example, when the total number of samples decided by the "samples" field is an exact multiple of the number of samples within each frame, which is decided by the "frame_length"
field, the multiple value becomes the total number of frames.  However, if the total number of samples decided by the "samples" field is not an exact multiple of the number of samples decided by the "frame_length" field, and a remainder (or rest) exist,
the total number of frames increases by "1" more than the multiple value.  Furthermore, the number of samples of the last frame (frlen_last) is decided as the remainder (or rest).  This indicates that only the number of samples of the last frame is
different from its previous frame.


 By defining a standardized rule between the encoder and the decoder, as described above, the encoder may freely decide and transmit the total number of samples ("samples" field) for each channel and the number of samples("frame_length" field)
within a frame of each channel.  Furthermore, the decoder may accurately decide, by using the above-described algorithm on the transmitted information, the number of frames for each channel that is to be used for decoding.


 Linear Prediction


 In the present invention, linear prediction is applied for the lossless audio coding.  The predictor 160 shown in FIG. 1 includes at least one or more filter coefficients so as to predict a current sample value from a previous sample value. 
Then, the second entropy coding part 180 performs entropy coding on a residual value corresponding to the difference between the predicted value and the original value.  Additionally, the predictor coefficient values for each block that are applied to
the predictor 160 are selected as optimum values from the coefficient estimating part 120.  Further, the predictor coefficient values are entropy coded by the first entropy coding part 140.  The data coded by the first entropy coding part and the second
entropy coding part 180 are inserted as part of the bitstream by the multiplexing part 190 and then transmitted.


 Hereinafter, the method of performing linear prediction according to the present invention will now be described in detail.


 Prediction with FIR Filters


 Linear prediction is used in many applications for speech and audio signal processing.  Hereinafter, an exemplary operation of the predictor 160 will be described based on Finite Impulse Response (FIR) filters.  However, it is apparent that this
example will not limit the scope of the present invention.


 The current sample of a time-discrete signal x(n) can be approximately predicted from previous samples x(n-k).  The prediction is given by the following equation.


 .function..times..times..function.  ##EQU00001##


 wherein K is the order of the predictor.  If the predicted samples are close to the original samples, the residual shown below: e(n)=x(n)-{circumflex over (x)}(n)


 has a smaller variance than x(n) itself, hence e(n) can be encoded more efficiently.


 The procedure of estimating the predictor coefficients from a segment of input samples, prior to filtering that segment is referred to as forward adaptation.  In this case, the coefficients should be transmitted.  On the other hand, if the
coefficients are estimated from previously processed segments or samples, e.g., from the residual, reference is made to backward adaptation.  The backward adaptation procedure has the advantage that no transmission of the coefficients is needed, since
the data required to estimate the coefficients is available to the decoder as well.


 Forward-adaptive prediction methods with orders around 10 are widely used in speech coding, and can be employed for lossless audio coding as well.  The maximum order of most forward-adaptive lossless prediction schemes is still rather small,
e.g., K=32.  An exception is the special 1-bit lossless codec for the Super Audio CD, which uses prediction orders of up to 128.


 On the other hand, backward-adaptive FIR filters with some hundred coefficients are commonly used in many areas, e.g., channel equalization and echo cancellation.  Most of these systems are based on the LMS algorithm or a variation thereof,
which has also been proposed for lossless audio coding.  Such LMS-based coding schemes with high orders are applicable since the predictor coefficients do not have to be transmitted as side information, thus their number does not contribute to the data
rate.  However, backward-adaptive codecs have the drawback that the adaptation has to be carried out both in the encoder and the decoder, making the decoder significantly more complex than in the forward-adaptive case.


 Forward-Adaptive Prediction


 As an exemplary embodiment of the present invention, forward adaptive prediction will be given as an example in the description set forth herein.  In forward-adaptive linear prediction, the optimal predictor coefficients h.sub.k (in terms of a
minimized variance of the residual) are usually estimated for each block by the coefficient estimating part 120 using the autocorrelation method or the covariance method.  The autocorrelation method, using the conventional Levinson-Durbin algorithm, has
the additional advantage of providing a simple means to iteratively adapt the order of the predictor.  Furthermore, the algorithm inherently calculates the corresponding parcor coefficients as well.


 Another aspect of forward-adaptive prediction is to determine a suitable prediction order.  Increasing the order decreases the variance of the prediction error, which leads to a smaller bit rate R.sub.e for the residual.  On the other hand, the
bit rate R.sub.c for the predictor coefficients will rise with the number of coefficients to be transmitted.  Thus, the task is to find the optimum order which minimizes the total bit rate.  This can be expressed by minimizing the equation below:,
R.sub.total(K)=R.sub.e(K)+R.sub.c(K), with respect to the prediction order K. As the prediction gain rises monotonically with higher orders, Re decreases with K. On the other hand R.sub.c rises monotonically with K, since an increasing number of
coefficients should be transmitted.


 The search for the optimum order can be carried out efficiently by the coefficient estimating part 120, which determines recursively all predictors with increasing order.  For each order, a complete set of predictor coefficients is calculated. 
Moreover, the variance .sigma..sub.e.sup.2 of the corresponding residual can be derived, resulting in an estimate of the expected bit rate for the residual.  Together with the bit rate for the coefficients, the total bit rate can be determined in each
iteration, i.e., for each prediction order.  The optimum order is found at the point where the total bit rate no longer decreases.


 While it is obvious from the above equation that the coefficient bit rate has a direct effect on the total bit rate, a slower increase of R.sub.c also allows to shift the minimum of R.sub.total to higher orders (wherein R.sub.e is smaller as
well), which would lead to better compression.  Hence, efficient yet accurate quantization of the predictor coefficients plays an important role in achieving maximum compression.


 Prediction Orders


 In the present invention, the prediction order K, which decides the number of predictor coefficients for linear prediction, is determined.  The prediction order K is also determined by the coefficient estimating part 120.  Herein, information on
the determined prediction order is included in the bitstream and then transmitted.


 The configuration syntax (Table 6) includes information related to the prediction order K. For example, a 1-bit to 10-bit "max_order" field corresponds to information indicating a maximum order value.  The highest value of the 1-bit to 10-bit
"max_order" field is K=1023 (e.g., 10-bit).  As another information related to the prediction order K, the configuration syntax (Table 6) includes a 1-bit "adapt_order" field, which indicates whether an optimum order for each block exists.  For example,
when "adapt_order=1", an optimum order should be provided for each block.  In a block_data syntax (Table 8), the optimum order is provided as a 1-bit to 10-bit "opt_order" field.  Further, when "adapt_order=0", a separate optimum order is not provided
for each block.  In this case, the "max_order" field becomes the final order applied to all of the blocks.


 The optimum order (opt_order) is decided based upon the value of max_order field and the size (N.sub.B) of the corresponding block.  More specifically, for example, when the max_order is decided as K.sub.max=10 and "adapt_order=1", the opt_order
for each block may be decided considering the size of the corresponding block.  In some case, the opt_order value being larger than max_order (K.sub.max=10) is possible.


 In particular, the present invention relates to higher prediction orders.  In the absence of hierarchical block switching, there may be a factor of 4 between the long and the short block length (e.g. 4096 & 1024 or 8192 & 2048), in accordance
with the embodiments.  On the other hand, in the embodiments where hierarchical block switching is implemented, this factor can be increased (e.g., up to 32), enabling a larger range (e.g., 16384 down to 512 or even 32768 to 1024 for high sampling
rates).


 In the embodiments where hierarchical block switching is implemented, in order to make better use of very long blocks, higher maximum prediction orders may be employed.  The maximum order may be K.sub.max=1023.  In the embodiments, K.sub.max may
be bound by the block length N.sub.B, for example, K.sub.max<N.sub.B/8 (e.g., K.sub.max=255 for N.sub.B=2048).  Therefore, using K.sub.max=1023 may require a block length of at least N.sub.B=8192.  In the embodiments, the "max_order" field in the
configuration syntax (Table 6) can be up to 10 bits and "opt_order" field in the block_data syntax (Table 8) can also be up to 10 bits.  The actual number of bits in a particular block may depend on the maximum order allowed for a block.  If the block is
short, a local prediction order may be smaller than a global prediction order.  Herein, the local prediction order is determined from considering the corresponding block length N.sub.B, and the global prediction order is determined from the "max_order"
K.sub.max in the configuration syntax.  For example, if K.sub.max=1023, but N.sub.B=2048, the "opt_order" field is determined on 8 bits (instead of 10) due to a local prediction order of 255.


 More specifically, the opt_order may be determined based on the following equation: opt_order=min (global prediction order, local prediction order); And.  the global and local prediction orders may be determined by: global prediction
order=ceil(log 2 (maximum prediction order+1)) local prediction order=max(ceil(log 2 ((Nb>>3)-1)),1)


 In the embodiments, data samples of the subdivided block from a channel are predicted.  A first sample of a current block is predicted using the last K samples of a previous block.  The K value is determined from the opt_order which is derived
from the above-described equation.


 If the current block is a first block of the channel, no samples from the previous block are used.  In this case, prediction with progressive order is employed.  For example, assuming that the opt_order value is K=5 for a corresponding block,
the first sample in the block does not perform prediction.  The second sample of the block uses the first sample of the block to perform the prediction (as like K=1), the third sample of the block uses the first and second samples of the block to perform
the prediction (as like K=2), etc. Therefore, starting from the sixth sample and for samples thereafter, prediction is performed according to the opt_order of K=5.  As described above, the prediction order increases progressively from K=1 to K=5.


 The above-described progressive order type of prediction is very advantageous when used in the random access frame.  Since the random access frame corresponds to a reference frame of the random access unit, the random access frame does not
perform prediction by using the previous frame sample.  Namely, this progressive prediction technique may be applied at the beginning of the random access frame.


 Quantization of Predictor Coefficients


 The above-described predictor coefficients are quantized in the quantizing part 130 of FIG. 1.  Direct quantization of the predictor coefficients h.sub.k is not very efficient for transmission, since even small quantization errors may result in
large deviations from the desired spectral characteristics of the optimum prediction filter.  For this reason, the quantization of predictor coefficients is based on the parcor (reflection) coefficients r.sub.k, which can be calculated by the coefficient
estimating part 120.  As described above, for example, the coefficient estimating part 120 is processed using the conventional Levinson-Durbin algorithm.


 The first two parcor coefficients (.gamma..sub.1 and .gamma..sub.2 correspondingly) are quantized by using the following functions: a.sub.1=.left brkt-bot.64(-1+ {square root over (2)} {square root over (.gamma..sub.1+1)}).right brkt-bot.;
a.sub.2=.left brkt-bot.64(-1+ {square root over (2)} {square root over (-.gamma..sub.2+1)}).right brkt-bot.; while the remaining coefficients are quantized using simple 7-bit uniform quantizers: a.sub.k=.left brkt-bot.64.gamma..sub.k.right brkt-bot.;
(k>2).


 In all cases the resulting quantized values a.sub.k are restricted to the range [-64,63].


 Entropy Coding


 As shown in FIG. 1, two types of entropy coding are applied in the present invention.  More specifically, the first entropy coding part 140 is used for coding the above-described predictor coefficients.  And, the second entropy coding part 180
is used for coding the above-described audio original samples and audio residual samples.  Hereinafter, the two types of entropy coding will now be described in detail.


First Entropy Coding of the Predictor Coefficient


 The related art Rice code is used as the first entropy coding method according to the present invention.  For example, transmission of the quantized coefficients a.sub.k is performed by producing residual values:
.delta..sub.k=a.sub.k-offset.sub.k, which, in turn, are encoded by using the first entropy coding part 140, e.g., the Rice code method.  The corresponding offsets and parameters of Rice code used in this process can be globally chosen from one of the
sets shown in Table 3, 4 and 5 below.  A table index (i.e., a 2-bit "coef_table") is indicated in the configuration syntax (Table 6).  If "coef_table=11", this indicates that no entropy coding is applied, and the quantized coefficients are transmitted
with 7 bits each.  In this case, the offset is always -64 in order to obtain unsigned values .delta..sub.k=a.sub.k+64 that are restricted to [0,127].  Conversely, if "coeff_table=00", Table 3 below is selected, and if "coeff_table=01", Table 4 below is
selected.  Finally, if "coeff_table=10", Table 5 is selected.


 When receiving the quantized coefficients in the decoder of FIG. 2, the first entropy decoding part 220 reconstructs the predictor coefficients by using the process that the residual values .delta..sub.k are combined with offsets to produce
quantized indices of parcor coefficients a.sub.k: a.sub.k=.delta..sub.k+offset.sub.k.


 Thereafter, the reconstruction of the first two coefficients (.gamma..sub.1 and .gamma..sub.2) is performed by using: par.sub.1=.left brkt-bot.{circumflex over (.gamma.)}.sub.12.sup.Q.right brkt-bot.=.GAMMA.(a.sub.1); par.sub.2=.left
brkt-bot.{circumflex over (.gamma.)}.sub.22.sup.Q.right brkt-bot.=-.GAMMA.(a.sub.2); wherein 2.sup.Q represents a constant (Q=20) scale factor required for integer representation of the reconstructed coefficients, and .GAMMA.(.) is an empirically
determined mapping table (not shown as the mapping table may vary with implementation).


 Accordingly, the three types of coefficient tables used for the first entropy coding are provided according to the sampling frequency.  For example, the sampling frequency may be divided to 48 kHz, 96 kHz, and 192 kHz.  Herein, each of the three
Tables 3, 4, and 5 is respectively provided for each sampling frequency.


 Instead of using a single table, one of three different tables can be chosen for the entire file.  The table should typically be chosen depending on the sampling rate.  For material with 44.1 kHz, the applicant of the present invention
recommends to use the 48 kHz table.  However, in general, the table can also be chosen by other criteria.


 TABLE-US-00004 TABLE 3 Rice code parameters used for encoding of quantized coefficients (48 kHz).  Coefficient # Offset Rice parameter 1 -52 4 2 -29 5 3 -31 4 4 19 4 5 -16 4 6 12 3 7 -7 3 8 9 3 9 -5 3 10 6 3 11 -4 3 12 3 3 13 -3 2 14 3 2 15 -2 2
16 3 2 17 -1 2 18 2 2 19 -1 2 20 2 2 2k - 1, k > 10 0 2 2k, k > 10 1 2


 TABLE-US-00005 TABLE 4 Rice code parameters used for encoding of quantized coefficients (96 kHz).  Coefficient # Offset Rice parameter 1 -58 3 2 -42 4 3 -46 4 4 37 5 5 -36 4 6 29 4 7 -29 4 8 25 4 9 -23 4 10 20 4 11 -17 4 12 16 4 13 -12 4 14 12 3
15 -10 4 16 7 3 17 -4 4 18 3 3 19 -1 3 20 1 3 2k - 1, k > 10 0 2 2k, k > 10 1 2


 TABLE-US-00006 TABLE 5 Rice code parameters used for encoding of quantized coefficients (192 kHz).  Coefficient # Offset Rice parameter 1 -59 3 2 -45 5 3 -50 4 4 38 4 5 -39 4 6 32 4 7 -30 4 8 25 3 9 -23 3 10 20 3 11 -20 3 12 16 3 13 -13 3 14 10
3 15 -7 3 16 3 3 17 0 3 18 -1 3 19 2 3 20 -1 2 2k - 1, k > 10 0 2 2k, k > 10 1 2


 Second Entropy Coding of the Residual


 The present invention contains two different modes of the coding method applied to the second entropy coding part 180 of FIG. 1, which will now be described in detail.


 In the simple mode, the residual values e(n) are entropy coded using Rice code.  For each block, either all values can be encoded using the same Rice code, or the block can be further divided into four parts, each encoded with a different Rice
code.  The indices of the applied codes are transmitted, as shown in FIG. 1.  Since there are different ways to determine the optimal Rice code for a given set of data, it is up to the encoder to select suitable codes depending upon the statistics of the
residual and the codeword generation shown below is only one of several possible Rice coding realizations.


 A Rice code is defined by a parameter s.gtoreq.0.  For a given value of s, each codeword consists of a p-bit prefix and an s-bit sub-code.  The prefix is signaled using p-1 "1"-bits and one "0"-bit, with p depending on the coded value.  For a
signal value x and s>0, p-1 is calculated as follows ("/" means integer division without remainder in the equations below):


 /.times..times..gtoreq./.times..times.< ##EQU00002## For s=0, we used a modified calculation:


 .times..times..times..gtoreq..times..times..times.< ##EQU00003## The sub-code for s>0 is calculated as follows:


 .function..times..times..times..gtoreq..times..times..times.< ##EQU00004## and for the ith sub-block.  For s=0 there is no sub-code but only the prefix, thus the prefix and the codeword are identical.  Table A and Table B below show examples
for the Rice code with s=4.  Table C shows the special Rice code with s=0.


 TABLE-US-00007 TABLE A Rice code with s = 4.  The xxxx bits contain the 4-bit sub-code s[i].  Values P Prefix Codeword -8 .  . . +7 1 0 0xxxx -16 .  . . -9; +8 .  . . +15 2 10 10xxxx -24 .  . . -17; +16 .  . . +23 3 110 110xxxx -32 .  . . -25;
+24 .  . . +31 4 1110 1110xxxx -40 .  . . -33; +32 .  . . +39 5 11110 11110xxxx


 TABLE-US-00008 TABLE B Sub-codes of the Rice code with s = 4 for the first three prefixes.  Values Values Values sub-code (p = 1) (p = 2) (p = 3) (xxxx) -8 -16 -24 0111 -7 -15 -23 0110 -6 -14 -22 0101 -5 -13 -21 0100 -4 -12 -20 0011 -3 -11 -19
0010 -2 -10 -18 0001 -1 -9 -17 0000 0 8 16 1000 1 9 17 1001 2 10 18 1010 3 11 19 1011 4 12 20 1100 5 13 21 1101 6 14 22 1110 7 15 23 1111


 TABLE-US-00009 TABLE C "Special" Rice code with s = 0.  Prefix and codeword are identical.  Values P Prefix Codeword 0 1 0 0 -1 2 10 10 +1 3 110 110 -2 4 1110 1110 +2 5 11110 11110


 Alternatively, the encoder can use a more complex and efficient coding scheme using BGMC mode.  In the BGMC mode, the encoding of residuals is accomplished by splitting the distribution in two categories.  The two types include residuals that
belong to a central region of the distribution, |e(n)|<e.sub.max, and residuals that belong to its tails.  The residuals in tails are simply re-centered (i.e., for e(n)>e.sub.max, e, (n)=e(n)-e.sub.max is provided) and encoded using Rice code as
described above.  However, in order to encode residuals in the center of the distribution, the BGMC first splits the residuals into LSB and MSB components, then the BGMC encodes MSBs using block Gilbert-Moore (arithmetic) codes.  And finally, the BGMC
transmits LSBs using direct fixed-lengths codes.  Both parameters e.sub.max and the number of directly transmitted LSBs may be selected such that they only slightly affect the coding efficiency of this scheme, while allowing the coding to be
significantly less complex.


 The configuration syntax (Table 6) and the block_data syntax (Table 8) according to the present invention include information related to coding of the Rice code and BGMC code.  The information will now be described in detail


 The configuration syntax (Table 6) first includes a 1-bit "bgmc_mode" field.  For example, "bgmc_mode=0" signifies the Rice code, and "bgmc_mode=1" signifies the BGMC code.  The configuration syntax (Table 6) also includes a 1-bit "sb_part"
field.  The "sb_part" field corresponds to information related to a method of partitioning a block into sub-block(s) and coding the partitioned sub-block.  Herein, the meaning of the "sb_part" field varies in accordance with the value of the "bgmc_mode"
field.


 For example, when "bgmc_mode=0", in other words when the Rice code is applied, "sb_part=0" signifies that the block is not partitioned into sub-blocks.  And, "sb_part=1" signifies that the block is partitioned at a 1:4 sub-block partition ratio. Additionally, when "bgmc_mode=1", in other words when the BGMC code is applied, "sb_part=0" signifies that the block is partitioned at a 1:4 sub-block partition ratio.  Alternatively, "sb_part=1" signifies that the block is partitioned at a 1:2:4:8
sub-block partition ratio.


 The block_data syntax (Table 8) for each block corresponding to the information included in the configuration syntax (Table 6) includes 0-bit to 2-bit variable "ec_sub" fields.  More specifically, the "ec_sub" field indicates the number of
sub-blocks existing in the actual corresponding block.  Herein, the meaning of the "ec_sub" field varies in accordance with the value of the "bgmc_mode"+"sb_part" fields within the configuration syntax (Table 6).


 For example, "bgmc_mode+sb_part=0" signifies that the Rice code does not configure the sub-block.  Herein, the "ec_sub" field is a 0-bit field, which signifies that no information is included.


 In addition, "bgmc_mode+sb_part=1" signifies that the Rice code or the BGMC code is used to partition the block to sub-blocks at a 1:4 rate.  Herein, only 1 bit is assigned to the "ec_sub" field.  For example, "ec_sub=0" indicates one sub-block
(i.e., the block is not partitioned to sub-blocks), and "ec_sub=1" indicates that 4 sub-blocks are configured.


 Furthermore, "bgmc_mode+sb_part=2" signifies that the BGMC code is used to partition the block to sub-blocks at a 1:2:4:8 rate.  Herein, 2 bits are assigned to the "ec_sub" field.  For example, "ec_sub=00" indicates one sub-block (i.e., the
block is not partitioned to sub-blocks), and, "ec_sub=01" indicates 2 sub-blocks.  Also, "ec_sub=10" indicates 4 sub-blocks, and "ec_sub=11" indicates 8 sub-blocks.


 The sub-blocks defined within each block as described above are coded by second entropy coding part 180 using a difference coding method.  An example of using the Rice code will now be described.  For each block of residual values, either all
values can be encoded using the same Rice code, or, if the "sb_part" field in the configuration syntax is set, the block can be partitioned into 4 sub-blocks, each encoded sub-block having a different Rice code.  In the latter case, the "ec_sub" field in
the block-data syntax (Table 8) indicates whether one or four blocks are used.


 While the code parameter s[i=0] of the first sub-block is directly transmitted with either 4 bits (resolution.ltoreq.16 bits) or 5 bits (resolution>16 bits), only the differences (s[i]-s[i-1]) of following parameters s[i>0] are
transmitted.  These differences are additionally encoded using appropriately chosen Rice codes again.  In this case, the Rice code parameter used for differences has the value of "0".  Alternatively, the selected entropy code may be BGMC code with a code
parameter value of "2".


 As will be appreciated, the decoder reads the bgmc_mode, sb_part and ec_sub information to process and decode a received audio signal.  For example, the audio signal including a block of audio data partitioned into N sub-blocks is received, and
this information is read.  Using this information, a plurality of code parameters s(0), s(1), .  . . s(N-1), respectively are generated.  Namely, the generation of code parameters includes detecting s(0) from the audio signal, detecting a difference
s(i)-s(i-1) from the audio signal for i=1 .  . . N-1, and calculating s(i) for i=1, .  . . , N-1 using s(0) and the detected differences.  The N sub-blocks may then be decoded using the entropy codes defined by the generated code parameters.


 Syntax


 According to the embodiment of the present invention, the syntax of the various information included in the audio bitstream are shown in the tables below.  Table 6 shows a configuration syntax for audio lossless coding.  The configuration syntax
may form a header periodically placed in the bitstream, may form a header of each frame; etc. Table 7 shows a frame-data syntax, and Table 8 shows a block-data syntax.


 TABLE-US-00010 TABLE 6 Configuration syntax.  Syntax Bits ALSSpecificConfig( ) { samp_freq; 32 samples; 32 channels; 16 file_type; 3 resolution; 3 floating; 1 msb_first; 1 frame_length; 16 random_access; 8 ra_flag; 2 adapt_order; 1 coef_table; 2
long_term_prediction; 1 max_order; 10 block_switching; 2 bgmc_mode; 1 sb_part; 1 joint_stereo; 1 mc_coding; 1 chan_config; 1 chan_sort; 1 crc_enabled; 1 RLSLMS 1 (reserved) 6 if (chan_config) { chan_config_info; 16 } if (chan_sort) { for (c = 0; c <
channels; c++) chan_pos[c]; 8 } header_size; 16 trailer_size; 16 orig_header[ ]; header_size * 8 orig_trailer[ ]; trailer_size * 8 if (crc_enabled) { crc; 32 } if ((ra_flag == 2) && (random_access > 0)) { for (f = 0; f < (samples - 1 /
frame_length) + 1; f++) { ra_unit_size 32 } } }


 TABLE-US-00011 TABLE 7 Frame_data syntax.  Syntax Bits frame_data( ) { if ((ra_flag == 1) && (frame_id % random_access == 0)) { ra_unit_size 32 } if (mc_coding && joint_stereo) { js_switch; 1 byte_align; } if (!mc_coding || js_switch) { for (c =
0; c < channels; c++) { if (block_switching) { bs_info; 8, 16, 32 } if (independent_bs) { for (b = 0; b < blocks; b++) { block_data(c); } } else{ for (b = 0; b < blocks; b++) { block_data(c); block_data(c+1); } c++; } } else{ if
(block_switching) { bs_info; 8, 16, 32 } for (b = 0; b < blocks; b++) { for (c = 0; c < channels; c++) { block_data(c); channel_data(c); } } } if (floating) { num_bytes_diff_float; 32 diff_float_data( ); } }


 TABLE-US-00012 TABLE 8 Block_data syntax.  Syntax Bits block_data( ) { block_type; 1 if (block_type == 0) { const_block; 1 js_block; 1 (reserved) 5 if (const_block == 1) { { if (resolution == 8) { const_val; 8 } else if (resolution == 16) {
const_val; 16 } else if (resolution == 24) { const_val; 24 } else { const_val; 32 } } } else { js_block; 1 if ((bgmc_mode == 0) && (sb_part == 0) { sub_blocks = 1; } else if ((bgmc_mode == 1) && (sb_part ==1) { ec_sub; 2 sub_blocks = 1 << ec_sub; }
else { ec_sub; 1 sub_blocks = (ec_sub == 1) ? 4 : 1; } if (bgmc_mode == 0) { for (k = 0; k < sub_blocks; k++) { s[k]; varies } } else { for (k = 0; k < sub_blocks; k++) { s[k],sx[k]; varies } } sb_length = block_length / sub_blocks; shift_lsbs; 1
if (shift_lsbs == 1) { shift_pos; 4 } if (!RLSLMS) { if (adapt_order == 1) { opt_order; 1...10 } for (p = 0; p < opt_order; p++) { quant_cof[p]; varies } }


 Compression Results


 In the following, the lossless audio codec is compared with two of the most popular programs for lossless audio compression: the open-source codec FLAC and the Monkey's Audio (MAC 3.97).  Herein, the open-source codec FLAC uses forward-adaptive
prediction, and the Monkey's Audio (MAC 3.97) is a backward-adaptive codec used as the current state-of-the-art algorithm in terms of compression.  Both codecs were run with options providing maximum compression (i.e., flac -8 and mac-c4000).  The
results for the encoder are determined for a medium compression level (with the prediction order restricted to K.sub.--60) and a maximum compression level (K.sub.--1023), both with random access of 500 ms.  The tests were conducted on a 1.7 GHz Pentium-M
system, with 1024 MB of memory.  The test comprises nearly 1 GB of stereo waveform data with sampling rates of 48, 96, and 192 kHz, and resolutions of 16 and 24 bits.


 Compression Ratio


 In the following, the compression ratio is defined as:


 .times.  ##EQU00005## wherein smaller values indicate better compression.  The results for the examined audio formats are shown in Table 9 (192 kHz material is not supported by the FLAC codec).


 TABLE-US-00013 TABLE 9 Comparison of average compression ratios for different audio formats (kHz/bits).  ALS ALS Format FLAC MAC medium maximum 48/16 48.6 45.3 45.5 44.7 48/24 68.4 63.2 63.3 62.7 96/24 56.7 48.1 46.5 46.2 192/24 -- 39.1 37.7
37.6 Total -- 48.9 48.3 47.8


 The results show that ALS at maximum level outperforms both FLAC and Monkey's Audio for all formats, but particularly for high-definition material (i.e., 96 kHz/24-bit and above).  Even at medium level, ALS delivers the best overall compression.


 Complexity


 The complexity of different codecs strongly depends on the actual implementation, particularly that of the encoder.  As mentioned above, the audio signal encoder of the present invention is an ongoing development.  Thus, we restrict our analysis
to the decoder, a simple C code implementation with no further optimizations.  The compressed data were generated by the currently best encoder implementation.  The average CPU load for real-time decoding of various audio formats, encoded at different
complexity levels, is shown in Table 10.  Even for maximum complexity, the CPU load of the decoder is only around 20-25%, which in return means that file based decoding is at least 4 to 5 times faster than real-time.


 TABLE-US-00014 TABLE 10 Average CPU load (percentage on a 1.7 GHz Pentium-M), depending on audio format (kHz/bits) and ALS encoder complexity.  ALS Format ALS low ALS medium maximum 48/16 1.6 4.9 18.7 48/24 1.8 5.8 19.6 96/24 3.6 12.0 23.8
192/24 6.7 22.8 26.7


 The codec is designed to offer a large range of complexity levels.  While the maximum level achieves the highest compression at the expense of slowest encoding and decoding speed, the faster medium level only slightly degrades compression, but
decoding is significantly less complex than for the maximum level (i.e., approximately 5% CPU load for 48 kHz material).  Using a low-complexity level (i.e., K.sub.--15, Rice coding) degrades compression by only 1 to 1.5% compared to the medium level,
but the decoder complexity is further reduced by a factor of three (i.e., less than 2% CPU load for 48 kHz material).  Thus, audio data can be decoded even on hardware with very low computing power.


 While the encoder complexity may be increased by both higher maximum orders and a more elaborate block switching algorithm (in accordance with the embodiments), the decoder may be affected by a higher average prediction order.


 The foregoing embodiments (e.g., hierarchical block switching) and advantages are merely examples and are not to be construed as limiting the appended claims.  The above teachings can be applied to other apparatuses and methods, as would be
appreciated by one of ordinary skill in the art.  Many alternatives, modifications, and variations will be apparent to those skilled in the art.


INDUSTRIAL APPLICABILITY


 It will be apparent to those skilled in the art that various modifications and variations can be made in the present invention without departing from the spirit or scope of the inventions.  For example, aspects and embodiments of the present
invention can be readily adopted in another audio signal codec like the lossy audio signal codec.  Thus, it is intended that the present invention covers the modifications and variations of this invention.


* * * * *























				
DOCUMENT INFO
Description: The present invention relates to a method for processing audio signal, and more particularly to a method and apparatus of encoding and decoding audio signal. The storage and replaying of audio signals has been accomplished in different ways in the past. For example, music and speech have been recorded and preserved by phonographic technology (e.g., record players), magnetic technology (e.g.,cassette tapes), and digital technology (e.g., compact discs). As audio storage technology progresses, many challenges need to be overcome to optimize the quality and storability of audio signals. For the archiving and broadband transmission of music signals, lossless reconstruction is becoming a more important feature than high efficiency in compression by means of perceptual coding as defined in MPEG standards such as MP3 or AAC. Although DVD audio and Super CD Audio include proprietary lossless compression schemes, there is a demand for an open and general compression scheme among content-holders and broadcasters. In response to this demand, a new lossless coding scheme hasbeen considered as an extension to the MPEG-4 Audio standard. Lossless audio coding permits the compression of digital audio data without any loss in quality due to a perfect reconstruction of the original signal.SUMMARY OF THE INVENTION The present invention relates to method of processing an audio signal. In one embodiment, at least one audio data frame having at least one channel is generated. Each channel is divided into a plurality of blocks. A sub-block partitioning scheme is selected, and a number of sub-blocks into which the block is tobe partitioned is selected. The selected number of sub-blocks is chosen from numbers of sub-blocks available for the selected sub-block partitioning scheme. The block of audio data is partitioned into sub-blocks according to the selected number ofsub-blocks, and the partitioned sub-blocks are coded according to a selected entropy coding scheme. In one embodime